The present invention relates to the transmission of data over error prone channels with fixed length data packages. It is especially suitable for perceptual audio coding.
Modern audio coding methods such as e.g. MPEG Layer 3, MPEG AAC or MPEG HE-AAC (MPEG=moving picture experts group, HE-AAC=high efficient advanced audio coding) are capable of reducing the data rate of digital audio signals by means of exploiting some psycho-acoustical properties of the human ear. Hereby a block of a fixed number of audio samples, called frame, is encoded to a compressed bit stream representation of this fixed time interval. The compressed audio frame will be transformed back to an audio sample representation in the decoder. Since the difficulty to encode an audio signal may vary for different audio frames, the well-known bit reservoir technique allows exchanging bits between the frames. Although the overall bit rate is constant, as a consequence the length of the frames in the bit stream is variable. The encoded frame has a part with side information containing essential information for the decoder to interpret the compressed data, followed by the compressed spectral data.
For transmission, the compressed audio frame has to be embedded into a transport format such as e.g. the ADTS (ADTS=audio data transport stream) or LOAS (LOAS=low overhead audio stream) transport formal for MPEG AAC. If there are errors in the transmission, it will be possible for the decoder to re-synchronize, due to sync-words, on the bit stream after the loss of one or more frames. Since in modern audio codecs, spectral data and parts of the side information is often entropy coded with code words of variable length such as e.g. Huffman coding in MPEG AAC, a single bit error is often sufficient for the decoder having to discard the whole frame and to mute the output signal or use some error concealment technique, e.g. noise insertion or interpolation between intact frames or a combination thereof. If longer regions of errors occur during the transmission, the decoder is still able to re-synchronize on the bit stream, but it does not have information about the number of frames that have been lost. In addition to the concealment of multiple frames, this can lead to audible time shift on the audio played back by the decoder or dropouts due to buffer over- or under-runs. Especially over error-prone channels, to keep a high quality of the transmitted audio signal, it is extremely important to have a sophisticated error-management available.
The invention is especially suited for the transmission over error prone channels with fixed length data segments. Because of the variable length of the frames, such as compressed audio frames, a new frame for a well-known transport format such as e.g. the already mentioned ADTS or LOAS formats usually starts at arbitrary positions of the fixed length data segment. Therefore, in case such a segment gets lost, which contains data of two consecutive frames, both frames will be corrupt and must be replaced by an error concealment strategy of the decoder.